----- Forwarded Message -----
From: John Donson <millim1974@xxxxxxxxx>
To: "alsa-user@xxxxxxxxxxxxxxxxxxxxx" <alsa-user@xxxxxxxxxxxxxxxxxxxxx>
Sent: Tuesday, November 15, 2011 8:25 PM
Subject: realtime adjustable delay with alsa in full-duplex capture and playback
From: John Donson <millim1974@xxxxxxxxx>
To: "alsa-user@xxxxxxxxxxxxxxxxxxxxx" <alsa-user@xxxxxxxxxxxxxxxxxxxxx>
Sent: Tuesday, November 15, 2011 8:25 PM
Subject: realtime adjustable delay with alsa in full-duplex capture and playback
Hi
I want to realize an application that features a constant but realtime adjustable delay between playback and capture with mmap alsa lib usage. The idea is to use up/down button with process messaging to fine tune sync between audio and video. I am using a commercial DVB-S reciever with AC3-OUT connected to the Linux PC SPDIF input of the soundcard and a subsequent TV for video display of the STB. Audio as about 500ms leading the corresponding image sequence. Which is good in general since I also plan to chain brutefir for DRC which gives we room for max 22000 samples FIR filter length/channel to be back in sync.
I know this can be accomplished by proper buffer length settings for a fixed delay. But it should be adjustable to
use it for other purpose as well.
Unfortunately I am not able to find a detailed diagram how alsa uses its ring-buffer and how interrupts do affect delay as well. Can I use snd_pcm_wait, snd_pcm_delay, snd_pcm_forward, and snd_pcm_rewind for that purpose and how? Is there a code fragment available?
Or in general, exists a code for SPDIF input to analog AC3 output with realtime adjustable delay? I managed easily to get it to work down to 3ms with arecord | ffmpeg spdif copy and ac3dec. For the application I am thinking to modify clalsadrv, ac3dec and irmp3 for my stand alone AC3 receiver Linux project. Better ideas welcome? But don't say JACK!
Regards
John
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